qmc_chan_get_byphandle() and the resource managed version retrieve a
channel from a simple phandle.
Extend the API and introduce qmc_chan_get_byphandles_index() and the
resource managed version in order to retrieve a channel from a phandle
list using the provided index to identify the phandle in the list.
Also update qmc_chan_get_byphandle() and the resource managed version to
use qmc_chan_get_byphandles_index() and so avoid code duplication.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-8-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Constraints are set by qmc_dai_startup(). These constraints are specific
to the interleaved mode.
With the future introduction of support for non-interleaved mode, a new
set of constraints will be set. To make the code clear and keep
qmc_dai_startup() simple, extract the current interleaved mode
constraints settings to a specific function.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-7-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Submitting data to QMC channels is done in several places: transfer
completions and DAI start. The operation done is simple and consist in
one function call.
With the future introduction of support for non-interleaved mode,
submitting data will be more complex.
To avoid copy/paste of code in several places, introduce
qmc_audio_pcm_{read,write}_submit() whose goal is to handle this
data submission.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-6-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The current QMC audio driver uses only one QMC channel per DAI. The
context used by QMC channel transfer (read and write) completion
routines does not contains any QMC channel and the only one available
per DAI is used to schedule the next transfer.
This works pretty well with only one QMC channel per DAI.
The future support for non-inlerleave mode will use several QMC channel
per DAI. In that case, QMC channel transfer completion routines need to
identify the QMC channel related to the completion.
In order to fill this lack, even if identifying the current QMC channel
among several QMC channels is not needed for the current code, add one
indirection level and introduce the qmc_dai_chan data structrure.
This structure contains the QMC channel involved in the completion and
refererences to the runtime context (capture and playback) used by the
DAI.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-5-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver mixes some internal values for channel DMA buffer handling
and PCM pointer handling. In the currently supported interleaved mode,
this mix does not lead to any issues but in order to prepare the
support for the non-interleaved mode, having them clearly separated will
ease the support and avoid additional computation to convert values used
in channel DMA buffer management in values usable for PCM pointer.
Use a specific set of variable for PCM pointer handling and an other set
for channel DMA buffer.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-4-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
./scripts/checkpatch.pl --strict --codespell detected several issues
when running on the fsl_qmc_audio.c file:
- CHECK: spaces preferred around that '*' (ctx:VxV)
- CHECK: Alignment should match open parenthesis
- CHECK: Comparison to NULL could be written "!prtd"
- CHECK: spaces preferred around that '/' (ctx:VxV)
- CHECK: Lines should not end with a '('
- CHECK: Please don't use multiple blank lines
Some of them are present several times.
Fix all of these issues without any functional changes.
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20240701113038.55144-3-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
These two commits set the upper limit of the Speaker Volume control
to +12dB instead of +100dB.
This should have been a simple 1-line change to the #define in the
header file, but only the HDA cs35l56 driver is using this define.
The ASoC cs35l56 driver was using hardcoded numbers instead of the
header defines.
So the first commit changes the ASoC driver to use the #defined
constants. The second commit corrects the value of the constant.
In the current flow all interrupts are disabled in runtime suspend
phase. However interrupts enablement only exists in fsl_xcvr_prepare().
After resume fsl_xcvr_prepare() may not be called so it will cause all
interrupts still disabled even if resume from suspend. Interrupts
should be explictily enabled after resume.
Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After
resume from suspend DPATH should be reset otherwise there'll be channel
swap issue.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20240628094354.780720-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Change CS35L56_MAIN_RENDER_USER_VOLUME_MAX to 48, to limit the maximum
value of the Speaker Volume control to +12dB. The minimum value is
unchanged so that the default 0dB has the same integer control value.
The original maximum of 400 (+100dB) was the largest value that can be
mathematically handled by the DSP. The actual maximum amplification is
+12dB.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240703095517.208077-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
Code used to create standalone and widget controls is mostly same, with
with exception that in standalone case dynamic object needs to be
registered and control created directly.
Following patches clean up and unify kcontrol creation code in topology
code.
Add a check to cs_dsp_coeff_write_ctrl() to abort if the control
is not writeable.
The cs_dsp code originated as an ASoC driver (wm_adsp) where all
controls were exported as ALSA controls. It relied on ALSA to
enforce the read-only permission. Now that the code has been
separated from ALSA/ASoC it must perform its own permission check.
This isn't currently causing any problems so there shouldn't be any
need to backport this. If the client of cs_dsp exposes the control as
an ALSA control, it should set permissions on that ALSA control to
protect it. The few uses of cs_dsp_coeff_write_ctrl() inside drivers
are for writable controls.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20240702110809.16836-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
simple-audio-mux is designed to be used generally, thus "Input 1" or
"Input 2" are used to selecting MUX input. This numbered inputs would work,
but might be not user friendly in some case, for example in case of system
hardware design has some clear labels.
Adds new "state-labels" property and enable to select MUX by own state names.
Original
> amixer set "MUX" "Input 1"
> amixer set "MUX" "Input 2"
Use mux-names
sound_mux: mux {
compatible = "simple-audio-mux";
mux-gpios = <...>;
=> state-labels = "Label_A", "Label_B";
};
> amixer set "MUX" "Label_A"
> amixer set "MUX" "Label_B"
Merge series from srinivas.kandagatla@linaro.org:
Existing way of allocating soundwire master ports on Qualcommm platforms is
dynamic, and in linear order starting from 1 to MAX_PORTS.
This will work as long as soundwire device ports are 1:1 mapped
linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these
are NOT mapped in that order.
The result of this is that only one speaker among the pair of speakers
is always silent, With recent changes for WSA codec to support codec
versions and along with these patches we are able to get all speakers
working on these SoCs.
bitfield.h is not explicitly included but it is required for FIELD_PREP
to be expanded by the preprocessor. If it is not implicitly included,
there will be a compiler error (as seen with ARCH=hexagon allmodconfig):
sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP);
| ^
sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8'
42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH'
65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */
| ^
sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4));
| ^
sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH'
70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */
| ^
Include bitfield.h explicitly, so that FIELD_PREP is always expanded,
clearing up the compiler error.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-include-for-field_prep-v1-1-0c5d7f71921b@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that ret may be used uninitialized in
lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with
CONFIG_WERROR):
sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized]
326 | "failed to init register map: %d\n", ret);
| ^~~
sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning
310 | int ret;
| ^
| = 0
1 error generated.
One solution would be a small refactoring of the second parameter in
dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in
the if statement. However, a nicer solution for debugging purposes,
which is the point of this statement, would be to use the '%pe'
specifier to symbolically print the error pointer value. Do so, which
eliminates the uninitialized use of ret, clearing up the warning.
Fixes: 0959de657a ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs")
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://patch.msgid.link/20240701-lpc32xx-asoc-fix-uninitialized-ret-v1-1-985d86189739@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
